[Freeswitch-users] 407 Proxy Authentication
David Ponzone
david.ponzone at gmail.com
Thu May 27 23:06:23 PDT 2010
If 63.79.178.192 is Verizon, It's Verizon sending the 407, so asking
you to authenticate.
David Ponzone Direction Technique
email: david.ponzone at ipeva.fr
tel: 01 74 03 18 97
gsm: 06 66 98 76 34
Service Client IPeva
tel: 0811 46 26 26
www.ipeva.fr - www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis
à l'intention exclusive de ses destinataires. Toute utilisation ou
diffusion non autorisée est interdite. Tout message électronique est
susceptible d'altération. IPeva décline toute responsabilité au
titre de ce message s'il a été altéré, déformé ou falsifié. Si
vous n'êtes pas destinataire de ce message, merci de le détruire
immédiatement et d'avertir l'expéditeur.
Le 28/05/2010 à 03:19, Michael De Lorenzo a écrit :
> I've tried updating our ACL conf, but that doesn't seem to help as I
> still get the 407 error (line 50 from the Gist below). Am I correct
> that the authentication is failing when Verizon attempts to contact
> our switch? Or is it an authentication failure when I hit Verizon?
>
> Here's the trace of the call attempt (I've replaced an IP address
> and phone number, so I do realize that they're not correct) - http://gist.github.com/416604
>
> send 1129 bytes to udp/[63.79.178.192]:5060 at 01:04:26.090037:
>
> ------------------------------------------------------------------------
> INVITE sip:17895551234 at 63.79.178.192 SIP/2.0
> Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c
> Max-Forwards: 70
> From: "C3 Mgmt" <sip:
> 9727289377 at 63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr
> To: <sip:17895551234 at 63.79.178.192>
> Call-ID: cd26b853-e497-122d-278a-000bdb94aab9
> CSeq: 131385997 INVITE
> Contact: <sip:gw+verizon at 0.0.0.0:5080;transport=udp;gw=verizon>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22
> 15:06:05 -0400
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, refer
> Privacy: none
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 291
> X-FS-Support: update_display
> P-Asserted-Identity: "C3 Mgmt" <sip:+19727289377 at 63.79.178.192>
>
> v=0
> o=FreeSWITCH 1274985280 1274985281 IN IP4 0.0.0.0
> s=FreeSWITCH
> c=IN IP4 0.0.0.0
> t=0 0
> m=audio 21600 RTP/AVP 0 8 3 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> ------------------------------------------------------------------------
> 2010-05-27 21:00:10.128406 [DEBUG] sofia.c:4168 Channel sofia/
> external/17895551234 entering state [calling][0]
> recv 296 bytes from udp/[63.79.178.192]:5060 at 01:00:10.192551:
>
> ------------------------------------------------------------------------
> SIP/2.0 100 Trying
> v: SIP/2.0/UDP
> 0.0.0.0:5080;rport;branch=z9hG4bKZc6Sjt81eNvHH;received=0.0.0.0
> f: "C3 Mgmt" <sip:
> 9727289377 at 63.79.178.192;transport=udp>;tag=S7QvyyUF1cpaD
> t: <sip:17895551234 at 63.79.178.192>
> i: 348b8efb-e497-122d-278a-000bdb94aab9
> CSeq: 131385870 INVITE
> l: 0
>
>
> ------------------------------------------------------------------------
> recv 425 bytes from udp/[63.79.178.192]:5060 at 01:00:10.192710:
>
> ------------------------------------------------------------------------
> SIP/2.0 407 Proxy Authentication Required
> v: SIP/2.0/UDP
> 0.0.0.0:5080;rport;branch=z9hG4bKZc6Sjt81eNvHH;received=0.0.0.0
> f: "C3 Mgmt" <sip:
> 9727289377 at 63.79.178.192;transport=udp>;tag=S7QvyyUF1cpaD
> t: <sip:17895551234 at 63.79.178.192>;tag=22703bab
> i: 348b8efb-e497-122d-278a-000bdb94aab9
> CSeq: 131385870 INVITE
> l: 0
> Proxy-Authenticate: DIGEST
> realm="WCOM",nonce="a586274e395fb9b6f66f1a7829b4531a.1275008409"
>
>
> ------------------------------------------------------------------------
> send 350 bytes to udp/[63.79.178.192]:5060 at 01:00:10.192977:
>
> ------------------------------------------------------------------------
> ACK sip:17895551234 at 63.79.178.192 SIP/2.0
> Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bKZc6Sjt81eNvHH
> Max-Forwards: 70
> From: "C3 Mgmt" <sip:
> 9727289377 at 63.79.178.192;transport=udp>;tag=S7QvyyUF1cpaD
> To: <sip:17895551234 at 63.79.178.192>;tag=22703bab
> Call-ID: 348b8efb-e497-122d-278a-000bdb94aab9
> CSeq: 131385870 ACK
> Content-Length: 0
>
>
> ------------------------------------------------------------------------
> 2010-05-27 21:00:10.193015 [DEBUG] sofia.c:4168 Channel sofia/
> external/17895551234 entering state [terminated][904]
> 2010-05-27 21:00:10.193015 [NOTICE] sofia.c:4804 Hangup sofia/
> external/17895551234 [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED]
> 2010-05-27 21:00:10.193015 [DEBUG] switch_channel.c:2117 Send signal
> sofia/external/17895551234 [KILL]
> 2010-05-27 21:00:10.193015 [DEBUG] switch_core_session.c:1022 Send
> signal sofia/external/17895551234 [BREAK]
> 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/17895551234) Running State Change CS_HANGUP
> 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:499
> (sofia/external/17895551234) State HANGUP
> 2010-05-27 21:00:10.193015 [DEBUG] mod_sofia.c:410 sofia/external/
> 17895551234 Overriding SIP cause 480 with 904 from the other leg
> 2010-05-27 21:00:10.193015 [DEBUG] mod_sofia.c:416 Channel sofia/
> external/17895551234 hanging up, cause: NORMAL_UNSPECIFIED
> 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:46
> sofia/external/17895551234 Standard HANGUP, cause: NORMAL_UNSPECIFIED
> 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:499
> (sofia/external/17895551234) State HANGUP going to sleep
> 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:333
> (sofia/external/17895551234) State Change CS_HANGUP -> CS_REPORTING
> 2010-05-27 21:00:10.193015 [DEBUG] switch_core_session.c:1022 Send
> signal sofia/external/17895551234 [BREAK]
> 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/17895551234) Running State Change CS_REPORTING
> 2010-05-27 21:00:10.193015 [DEBUG] switch_ivr_originate.c:3228
> Originate Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED]
> 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:590
> (sofia/external/17895551234) State REPORTING
>
>
> freeswitch at govinteract-fs-dev-2> 2010-05-27 21:00:10.299296 [DEBUG]
> switch_core_state_machine.c:53 sofia/external/17895551234 Standard
> REPORTING, cause: NORMAL_UNSPECIFIED
> 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:590
> (sofia/external/17895551234) State REPORTING going to sleep
> 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:327
> (sofia/external/17895551234) State Change CS_REPORTING -> CS_DESTROY
> 2010-05-27 21:00:10.299296 [DEBUG] switch_core_session.c:1022 Send
> signal sofia/external/17895551234 [BREAK]
> 2010-05-27 21:00:10.299296 [DEBUG] switch_core_session.c:1165
> Session 6 (sofia/external/17895551234) Locked, Waiting on external
> entities
> 2010-05-27 21:00:10.299296 [NOTICE] switch_core_session.c:1183
> Session 6 (sofia/external/17895551234) Ended
> 2010-05-27 21:00:10.299296 [NOTICE] switch_core_session.c:1185 Close
> Channel sofia/external/17895551234 [CS_DESTROY]
> 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:428
> (sofia/external/17895551234) Running State Change CS_DESTROY
> 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:439
> (sofia/external/17895551234) State DESTROY
> 2010-05-27 21:00:10.299296 [DEBUG] mod_sofia.c:343 sofia/external/
> 17895551234 SOFIA DESTROY
> 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:60
> sofia/external/17895551234 Standard DESTROY
> 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:439
> (sofia/external/17895551234) State DESTROY going to sleep
>
> freeswitch at govinteract-fs-dev-2> lua test1.lua
> 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885
> variable string 0 = [sip_cid_type=pid]
> 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885
> variable string 1 = [origination_caller_id_name=C3 Mgmt]
> 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885
> variable string 2 = [origination_caller_id_number=+19727289377]
> 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885
> variable string 3 = [ignore_early_media=true]
> 2010-05-27 21:04:26.088360 [NOTICE] switch_channel.c:669 New Channel
> sofia/external/17895551234 [f5c8454e-69f4-11df-b69d-099cfc924087]
> 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:3444 (sofia/external/
> 17895551234) State Change CS_NEW -> CS_INIT
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_session.c:1022 Send
> signal sofia/external/17895551234 [BREAK]
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/17895551234) Running State Change CS_INIT
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:338
> (sofia/external/17895551234) State INIT
> 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:83 sofia/external/
> 17895551234 SOFIA INIT
> 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:117 (sofia/external/
> 17895551234) State Change CS_INIT -> CS_ROUTING
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_session.c:1022 Send
> signal sofia/external/17895551234 [BREAK]
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:338
> (sofia/external/17895551234) State INIT going to sleep
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/17895551234) Running State Change CS_ROUTING
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/17895551234) State ROUTING
> 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:140 sofia/external/
> 17895551234 SOFIA ROUTING
> 2010-05-27 21:04:26.088360 [DEBUG] switch_ivr_originate.c:66 (sofia/
> external/17895551234) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_session.c:1022 Send
> signal sofia/external/17895551234 [BREAK]
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:341
> (sofia/external/17895551234) State ROUTING going to sleep
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/17895551234) Running State Change CS_CONSUME_MEDIA
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:360
> (sofia/external/17895551234) State CONSUME_MEDIA
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:360
> (sofia/external/17895551234) State CONSUME_MEDIA going to sleep
> send 1129 bytes to udp/[63.79.178.192]:5060 at 01:04:26.090037:
>
> ------------------------------------------------------------------------
> INVITE sip:17895551234 at 63.79.178.192 SIP/2.0
> Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c
> Max-Forwards: 70
> From: "C3 Mgmt" <sip:
> 9727289377 at 63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr
> To: <sip:17895551234 at 63.79.178.192>
> Call-ID: cd26b853-e497-122d-278a-000bdb94aab9
> CSeq: 131385997 INVITE
> Contact: <sip:gw+verizon at 0.0.0.0:5080;transport=udp;gw=verizon>
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22
> 15:06:05 -0400
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, refer
> Privacy: none
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 291
> X-FS-Support: update_display
> P-Asserted-Identity: "C3 Mgmt" <sip:+19727289377 at 63.79.178.192>
>
> v=0
> o=FreeSWITCH 1274985280 1274985281 IN IP4 0.0.0.0
> s=FreeSWITCH
> c=IN IP4 0.0.0.0
> t=0 0
> m=audio 23386 RTP/AVP 0 8 3 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> ------------------------------------------------------------------------
> 2010-05-27 21:04:26.089756 [DEBUG] sofia.c:4168 Channel sofia/
> external/17895551234 entering state [calling][0]
> recv 425 bytes from udp/[63.79.178.192]:5060 at 01:04:26.167603:
>
> ------------------------------------------------------------------------
> SIP/2.0 407 Proxy Authentication Required
> v: SIP/2.0/UDP
> 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c;received=0.0.0.0
> f: "C3 Mgmt" <sip:
> 9727289377 at 63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr
> t: <sip:17895551234 at 63.79.178.192>;tag=41e44fd5
> i: cd26b853-e497-122d-278a-000bdb94aab9
> CSeq: 131385997 INVITE
> l: 0
> Proxy-Authenticate: DIGEST
> realm="WCOM",nonce="43ea850af91d419e72d85c0375a60237.1275008665"
>
>
> ------------------------------------------------------------------------
> send 350 bytes to udp/[63.79.178.192]:5060 at 01:04:26.167913:
>
> ------------------------------------------------------------------------
> ACK sip:17895551234 at 63.79.178.192 SIP/2.0
> Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c
> Max-Forwards: 70
> From: "C3 Mgmt" <sip:
> 9727289377 at 63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr
> To: <sip:17895551234 at 63.79.178.192>;tag=41e44fd5
> Call-ID: cd26b853-e497-122d-278a-000bdb94aab9
> CSeq: 131385997 ACK
> Content-Length: 0
>
>
> ------------------------------------------------------------------------
> send 1359 bytes to udp/[63.79.178.192]:5060 at 01:04:26.168477:
>
> ------------------------------------------------------------------------
> INVITE sip:17895551234 at 63.79.178.192 SIP/2.0
> Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr
> Max-Forwards: 70
> From: "C3 Mgmt" <sip:
> 9727289377 at 63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr
> To: <sip:17895551234 at 63.79.178.192>
> Call-ID: cd26b853-e497-122d-278a-000bdb94aab9
> CSeq: 131385998 INVITE
> Contact: <sip:gw+verizon at 0.0.0.0:5080;transport=udp;gw=verizon>
> Expires: 3600
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22
> 15:06:05 -0400
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, refer
> Proxy-Authorization: Digest username="9727289377", realm="WCOM",
> nonce="43ea850af91d419e72d85c0375a60237.1275008665", algorithm=MD5,
> uri="sip:17895551234 at 63.79.178.192",
> response="1871a9dbd248f92b65df2af84c6057f8"
> Privacy: none
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 291
> X-FS-Support: update_display
> P-Asserted-Identity: "C3 Mgmt" <sip:+19727289377 at 63.79.178.192>
>
> v=0
> o=FreeSWITCH 1274985280 1274985281 IN IP4 0.0.0.0
> s=FreeSWITCH
> c=IN IP4 0.0.0.0
> t=0 0
> m=audio 23386 RTP/AVP 0 8 3 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> ------------------------------------------------------------------------
> 2010-05-27 21:04:26.168049 [DEBUG] sofia.c:4168 Channel sofia/
> external/17895551234 entering state [calling][0]
> recv 296 bytes from udp/[63.79.178.192]:5060 at 01:04:26.231231:
>
> ------------------------------------------------------------------------
> SIP/2.0 100 Trying
> v: SIP/2.0/UDP
> 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr;received=0.0.0.0
> f: "C3 Mgmt" <sip:
> 9727289377 at 63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr
> t: <sip:17895551234 at 63.79.178.192>
> i: cd26b853-e497-122d-278a-000bdb94aab9
> CSeq: 131385998 INVITE
> l: 0
>
>
> ------------------------------------------------------------------------
> recv 425 bytes from udp/[63.79.178.192]:5060 at 01:04:26.231408:
>
> ------------------------------------------------------------------------
> SIP/2.0 407 Proxy Authentication Required
> v: SIP/2.0/UDP
> 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr;received=0.0.0.0
> f: "C3 Mgmt" <sip:
> 9727289377 at 63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr
> t: <sip:17895551234 at 63.79.178.192>;tag=1da20439
> i: cd26b853-e497-122d-278a-000bdb94aab9
> CSeq: 131385998 INVITE
> l: 0
> Proxy-Authenticate: DIGEST
> realm="WCOM",nonce="43ea850af91d419e72d85c0375a60237.1275008665"
>
>
> ------------------------------------------------------------------------
> send 350 bytes to udp/[63.79.178.192]:5060 at 01:04:26.231667:
>
> ------------------------------------------------------------------------
> ACK sip:17895551234 at 63.79.178.192 SIP/2.0
> Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr
> Max-Forwards: 70
> From: "C3 Mgmt" <sip:
> 9727289377 at 63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr
> To: <sip:17895551234 at 63.79.178.192>;tag=1da20439
> Call-ID: cd26b853-e497-122d-278a-000bdb94aab9
> CSeq: 131385998 ACK
> Content-Length: 0
>
>
> ------------------------------------------------------------------------
> 2010-05-27 21:04:26.231294 [DEBUG] sofia.c:4168 Channel sofia/
> external/17895551234 entering state [terminated][904]
> 2010-05-27 21:04:26.231294 [NOTICE] sofia.c:4804 Hangup sofia/
> external/17895551234 [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED]
> 2010-05-27 21:04:26.231294 [DEBUG] switch_channel.c:2117 Send signal
> sofia/external/17895551234 [KILL]
> 2010-05-27 21:04:26.231294 [DEBUG] switch_core_session.c:1022 Send
> signal sofia/external/17895551234 [BREAK]
> 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/17895551234) Running State Change CS_HANGUP
> 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:499
> (sofia/external/17895551234) State HANGUP
> 2010-05-27 21:04:26.232334 [DEBUG] mod_sofia.c:410 sofia/external/
> 17895551234 Overriding SIP cause 480 with 904 from the other leg
> 2010-05-27 21:04:26.232334 [DEBUG] mod_sofia.c:416 Channel sofia/
> external/17895551234 hanging up, cause: NORMAL_UNSPECIFIED
> 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:46
> sofia/external/17895551234 Standard HANGUP, cause: NORMAL_UNSPECIFIED
> 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:499
> (sofia/external/17895551234) State HANGUP going to sleep
> 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:333
> (sofia/external/17895551234) State Change CS_HANGUP -> CS_REPORTING
> 2010-05-27 21:04:26.232334 [DEBUG] switch_core_session.c:1022 Send
> signal sofia/external/17895551234 [BREAK]
> 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:314
> (sofia/external/17895551234) Running State Change CS_REPORTING
> 2010-05-27 21:04:26.232334 [DEBUG] switch_ivr_originate.c:3228
> Originate Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED]
> 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:590
> (sofia/external/17895551234) State REPORTING
>
>
> freeswitch at govinteract-fs-dev-2> 2010-05-27 21:04:26.260365 [DEBUG]
> switch_core_state_machine.c:53 sofia/external/17895551234 Standard
> REPORTING, cause: NORMAL_UNSPECIFIED
> 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:590
> (sofia/external/17895551234) State REPORTING going to sleep
> 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:327
> (sofia/external/17895551234) State Change CS_REPORTING -> CS_DESTROY
> 2010-05-27 21:04:26.260365 [DEBUG] switch_core_session.c:1022 Send
> signal sofia/external/17895551234 [BREAK]
> 2010-05-27 21:04:26.260365 [DEBUG] switch_core_session.c:1165
> Session 7 (sofia/external/17895551234) Locked, Waiting on external
> entities
> 2010-05-27 21:04:26.260365 [NOTICE] switch_core_session.c:1183
> Session 7 (sofia/external/17895551234) Ended
> 2010-05-27 21:04:26.260365 [NOTICE] switch_core_session.c:1185 Close
> Channel sofia/external/17895551234 [CS_DESTROY]
> 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:428
> (sofia/external/17895551234) Running State Change CS_DESTROY
> 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:439
> (sofia/external/17895551234) State DESTROY
> 2010-05-27 21:04:26.260365 [DEBUG] mod_sofia.c:343 sofia/external/
> 17895551234 SOFIA DESTROY
> 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:60
> sofia/external/17895551234 Standard DESTROY
> 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:439
> (sofia/external/17895551234) State DESTROY going to sleep
> Here's how I've edited our acl.conf.xml file:
>
> <configuration name="acl.conf" description="Network Lists">
> <network-lists>
> <!--
> These ACL's are automatically created on startup.
>
> rfc1918.auto - RFC1918 Space
> nat.auto - RFC1918 Excluding your local lan.
> localnet.auto - ACL for your local lan.
> loopback.auto - ACL for your local lan.
> -->
>
> <list name="lan" default="allow">
> <node type="deny" cidr="192.168.42.0/24"/>
> <node type="allow" cidr="192.168.42.42/32"/>
> </list>
> <list name="verizon" default="deny">
> <node type="allow" cidr="63.79.178.192"/>
> </list>
> .... truncated, the rest is the default file ....
>
> Any help would be greatly appreciated!
>
>
> 2010/5/27 <freeswitch-users-request at lists.freeswitch.org>
> Send FreeSWITCH-users mailing list submissions to
> freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
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>
> You can reach the person managing the list at
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>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
> 1. Re: Don't work playback after bypass media mode. (Sergey
> Scheglov)
> 2. Re: Don't work playback after bypass media mode. (Sergey
> Scheglov)
> 3. Re: Don't work playback after bypass media mode.
> (Anthony Minessale)
> 4. Re: Don't work playback after bypass media mode. (Sergey
> Scheglov)
> 5. Re: Don't work playback after bypass media mode. (Sergey
> Scheglov)
>
>
> ---------- Forwarded message ----------
> From: Sergey Scheglov <sid at eltc.ru>
> To: freeswitch-users at lists.freeswitch.org
> Date: Fri, 28 May 2010 01:31:26 +0700
> Subject: Re: [Freeswitch-users] Don't work playback after bypass
> media mode.
> Hi, Anthony.
>
>
>
> Thank's for reply.
>
>
>
> You wrote 27 may 2010 г., 20:34:15:
>
>
>
> > playback executes on line 87 of your trace.
>
>
>
> yes, but executed by log after 30 seconds after line 84 and
> immediately hangup.
>
> Note, duration wav file - 4 sec.
>
>
>
> > If you do not hear the audio, It means the re-establishment of
> media fails somehow based on your topology or
>
> the phone you are on does not support early media,
>
>
>
> I don't hear the audio, because FS don't send RTP traffic to my
> phone in early media mode (checked sniffers) in my case.
>
> If dialplans is:
>
> <extension name="fail_balance">
>
> <condition field="${blabla}" expression="^\-1$">
>
> <action application="pre_answer"/>
>
> <action application="sleep" data="1000"/>
>
> <action application="playback" data="elight/neg_balance.wav"/>
>
> <action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
>
> </condition>
>
> </extension>
>
> then work's fine, no problems (bypass_media=true not set).
>
>
>
> > change your pre-answer before the playback to answer to verify.
>
>
>
> If set answer, problem disappears. But answer it's not for my case.
>
>
>
> > Try adding sofia profile internal siptrace on to see the sip
> traffic too.
>
>
>
> Call log with sip trace http://pastebin.freeswitch.org/13065
>
>
>
> Thanks again :)
>
>
>
> --
>
> Regard's
>
> Sergey Scheglov
>
>
>
> ---------- Forwarded message ----------
> From: Sergey Scheglov <sid at eltc.ru>
> To: Anthony Minessale <freeswitch-users at lists.freeswitch.org>
> Date: Fri, 28 May 2010 01:35:47 +0700
> Subject: Re: [Freeswitch-users] Don't work playback after bypass
> media mode.
> Hi, Anthony.
>
>
>
> You wrote 27 may 2010 г., 20:34:15:
>
>
>
>
> playback executes on line 87 of your trace.
>
>
>
> If you do not hear the audio, It means the re-establishment of media
> fails somehow based on your topology or
>
> the phone you are on does not support early media, change your pre-
> answer before the playback to answer to verify.
>
>
>
> Try adding sofia profile internal siptrace on to see the sip traffic
> too.
>
>
>
>
>
>
>
> > playback executes on line 87 of your trace.
>
>
>
> yes, but executed by log after 30 seconds after line 84 and
> immediately hangup.
>
> Note, duration wav file - 4 sec.
>
>
>
> > If you do not hear the audio, It means the re-establishment of
> media fails somehow based on your topology or
>
> the phone you are on does not support early media,
>
>
>
> I don't hear the audio, because FS don't send RTP traffic to my
> phone in early media mode (checked sniffers) in my case.
>
> If dialplans is:
>
> <extension name="fail_balance">
>
> <condition field="${blabla}" expression="^\-1$">
>
> <action application="pre_answer"/>
>
> <action application="sleep" data="1000"/>
>
> <action application="playback" data="elight/neg_balance.wav"/>
>
> <action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
>
> </condition>
>
> </extension>
>
> then work's fine, no problems (bypass_media=true not set).
>
>
>
> > change your pre-answer before the playback to answer to verify.
>
>
>
> If set answer, problem disappears. But answer it's not for my case.
>
>
>
> > Try adding sofia profile internal siptrace on to see the sip
> traffic too.
>
>
>
> Call log with sip trace http://pastebin.freeswitch.org/13065
>
>
>
> Thanks again :)
>
>
>
>
>
> --
>
> Regard's
>
> Sergey Scheglov
>
>
>
> ---------- Forwarded message ----------
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Thu, 27 May 2010 13:46:05 -0500
> Subject: Re: [Freeswitch-users] Don't work playback after bypass
> media mode.
> it's against the SIP spec to re-negotiate media before you have
> answered.
> the transaction to negotiate the SDP from the original call has not
> completed so it's illegal to send a re-invite.
>
> instead you should use bypass_media_after_bridge=true so the bypass
> only happens when the bridge works.
>
>
> 2010/5/27 Sergey Scheglov <sid at eltc.ru>
> Hi, Anthony.
>
>
>
> You wrote 27 may 2010 г., 20:34:15:
>
>
>
>
> playback executes on line 87 of your trace.
>
>
>
> If you do not hear the audio, It means the re-establishment of media
> fails somehow based on your topology or
>
> the phone you are on does not support early media, change your pre-
> answer before the playback to answer to verify.
>
>
>
> Try adding sofia profile internal siptrace on to see the sip traffic
> too.
>
>
>
>
>
>
>
> > playback executes on line 87 of your trace.
>
>
>
> yes, but executed by log after 30 seconds after line 84 and
> immediately hangup.
>
> Note, duration wav file - 4 sec.
>
>
>
> > If you do not hear the audio, It means the re-establishment of
> media fails somehow based on your topology or
>
> the phone you are on does not support early media,
>
>
>
> I don't hear the audio, because FS don't send RTP traffic to my
> phone in early media mode (checked sniffers) in my case.
>
> If dialplans is:
>
> <extension name="fail_balance">
>
> <condition field="${blabla}" expression="^\-1$">
>
> <action application="pre_answer"/>
>
> <action application="sleep" data="1000"/>
>
> <action application="playback" data="elight/neg_balance.wav"/>
>
> <action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
>
> </condition>
>
> </extension>
>
> then work's fine, no problems (bypass_media=true not set).
>
>
>
> > change your pre-answer before the playback to answer to verify.
>
>
>
> If set answer, problem disappears. But answer it's not for my case.
>
>
>
> > Try adding sofia profile internal siptrace on to see the sip
> traffic too.
>
>
>
> Call log with sip trace http://pastebin.freeswitch.org/13065
>
>
>
> Thanks again :)
>
>
>
>
>
> --
>
> Regard's
>
> Sergey Scheglov
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> ---------- Forwarded message ----------
> From: Sergey Scheglov <sid at eltc.ru>
> To: Anthony Minessale <freeswitch-users at lists.freeswitch.org>
> Date: Fri, 28 May 2010 01:49:04 +0700
> Subject: Re: [Freeswitch-users] Don't work playback after bypass
> media mode.
> Hi, Anthony.
>
> Thank's for reply.
>
> You wrote 27 may 2010 г., 20:34:15:
>
> > playback executes on line 87 of your trace.
>
> yes, but executed by log after 30 seconds after line 84 and
> immediately hangup.
> Note, duration wav file - 4 sec.
>
> > If you do not hear the audio, It means the re-establishment of
> media fails
> > somehow based on your topology or
> > the phone you are on does not support early media, change your pre-
> answer
> > before the playback to answer to verify.
>
> I don't hear the audio, because FS don't send RTP traffic to my
> phone in early media mode (checked sniffers) in my case.
> If dialplans is:
> <extension name="fail_balance">
> <condition field="${blabla}" expression="^\-1$">
> <action application="pre_answer"/>
> <action application="sleep" data="1000"/>
> <action application="playback" data="elight/neg_balance.wav"/>
> <action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
> </condition>
> </extension>
> then work's fine, no problems (bypass_media=true not set).
>
> If set answer, problem disappears. But answer it's not for my case.
>
> > Try adding sofia profile internal siptrace on to see the sip
> traffic too.
>
> Call log with sip trace http://pastebin.freeswitch.org/13065
>
> Thanks again :)
>
> Best regard's
> Sergey Scheglov
>
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Sergey Scheglov <sid at eltc.ru>
> To: Anthony Minessale <freeswitch-users at lists.freeswitch.org>
> Date: Fri, 28 May 2010 02:12:28 +0700
> Subject: Re: [Freeswitch-users] Don't work playback after bypass
> media mode.
> Hi, Anthony.
>
> Thanks, thanks, thanks ))
> You wrote 28 may 2010., 1:46:05:
>
> > it's against the SIP spec to re-negotiate media before you have
> answered.
> > the transaction to negotiate the SDP from the original call has not
> > completed so it's illegal to send a re-invite.
>
> Do I understand correctly that the dialplan is not correct?
>
> > instead you should use bypass_media_after_bridge=true so the
> bypass only
> > happens when the bridge works.
>
> As I wrote in first message:
> "If set bypass_media_after_bridge instead bypass_media, then works
> fine, BUT changing codec negotiation." And two phones that support
> codecs, which not present in internal profile, not be able to call
> each other.
>
> Best regards
> Sergey Scheglov
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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